Page 1 of 1

Calls Failing With 408, Request Timeout.

PostPosted: Tue Aug 29, 2017 11:20 am
by amitiyer
Hello All,

I need help, i am trying to make a call to my carrier and when i do the call fails with 408, Request Timeout everything. Below are the sip.conf and dialplan below. Please also check the sip debug file attached.

Please help me here.

Vicidial Version : VERSION: 2.14-626a / BUILD: 170825-1708
Asterisk Version : 11.22.0

sip.conf
[newcarrier]
disallow=all
allow=ulaw
dtmf=rfc2833
nat=no
type=friend
host=carrierIP
allow=alaw
context=trunkinbound
insecure=invite,port
bindport=5060
qualify=yes


Dialplan :
exten => _9X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9X.,n,Dial(sip/${EXTEN:1}@newcarrier,55,o)
exten => _9X.,n,Hangup()



SIP DEBUG
Code: Select all
SIP Debugging enabled

<--- SIP read from UDP:(Origination IP):8364 --->
INVITE sip:9919033016760@(Vicidial Server IP) SIP/2.0
Via: SIP/2.0/UDP (Origination IP):8364;branch=z9hG4bK-d8754z-cc00a3584e400b3f-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:8001@(Origination IP):8364>
To: "9919033016760"<sip:9919033016760@(Vicidial Server IP)>
From: "8001"<sip:8001@(Vicidial Server IP)>;tag=f37ee657
Call-ID: MGYzMDJiMTY3YmUzNGIwZDkyYjFjYTU2NjFmM2NiMTI.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: eyeBeam release 1102q stamp 51814
Content-Length: 289

v=0
o=- 9 2 IN IP4 (Origination IP)
s=CounterPath eyeBeam 1.5
c=IN IP4 (Origination IP)
t=0 0
m=audio 33798 RTP/AVP 0 8 101
a=alt:1 2 : fwXYnWQI QO8ESQ1f 192.168.56.1 33798
a=alt:2 1 : UvciliRu 3lOWh2Bo 192.168.4.108 33798
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv
<------------->
--- (12 headers 11 lines) ---
Sending to (Origination IP):8364 (NAT)
Sending to (Origination IP):8364 (NAT)
Using INVITE request as basis request - MGYzMDJiMTY3YmUzNGIwZDkyYjFjYTU2NjFmM2NiMTI.
Found peer '8001' for '8001' from (Origination IP):8364

<--- Reliably Transmitting (NAT) to (Origination IP):8364 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP (Origination IP):8364;branch=z9hG4bK-d8754z-cc00a3584e400b3f-1---d8754z-;received=(Origination IP);rport=8364
From: "8001"<sip:8001@(Vicidial Server IP)>;tag=f37ee657
To: "9919033016760"<sip:9919033016760@(Vicidial Server IP)>;tag=as4f410be6
Call-ID: MGYzMDJiMTY3YmUzNGIwZDkyYjFjYTU2NjFmM2NiMTI.
CSeq: 1 INVITE
Server: Asterisk PBX 11.22.0-vici
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7f27116a"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'MGYzMDJiMTY3YmUzNGIwZDkyYjFjYTU2NjFmM2NiMTI.' in 23872 ms (Method: INVITE)

<--- SIP read from UDP:(Origination IP):8364 --->
ACK sip:9919033016760@(Vicidial Server IP) SIP/2.0
Via: SIP/2.0/UDP (Origination IP):8364;branch=z9hG4bK-d8754z-cc00a3584e400b3f-1---d8754z-;rport
To: "9919033016760"<sip:9919033016760@(Vicidial Server IP)>;tag=as4f410be6
From: "8001"<sip:8001@(Vicidial Server IP)>;tag=f37ee657
Call-ID: MGYzMDJiMTY3YmUzNGIwZDkyYjFjYTU2NjFmM2NiMTI.
CSeq: 1 ACK
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

<--- SIP read from UDP:(Origination IP):8364 --->
INVITE sip:9919033016760@(Vicidial Server IP) SIP/2.0
Via: SIP/2.0/UDP (Origination IP):8364;branch=z9hG4bK-d8754z-3b6e8b49f82c6334-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:8001@(Origination IP):8364>
To: "9919033016760"<sip:9919033016760@(Vicidial Server IP)>
From: "8001"<sip:8001@(Vicidial Server IP)>;tag=f37ee657
Call-ID: MGYzMDJiMTY3YmUzNGIwZDkyYjFjYTU2NjFmM2NiMTI.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: eyeBeam release 1102q stamp 51814
Authorization: Digest username="8001",realm="asterisk",nonce="7f27116a",uri="sip:9919033016760@(Vicidial Server IP)",response="e4b958c52b6ec45ff67ab995c67ae0b6",algorithm=MD5
Content-Length: 289

v=0
o=- 9 2 IN IP4 (Origination IP)
s=CounterPath eyeBeam 1.5
c=IN IP4 (Origination IP)
t=0 0
m=audio 33798 RTP/AVP 0 8 101
a=alt:1 2 : fwXYnWQI QO8ESQ1f 192.168.56.1 33798
a=alt:2 1 : UvciliRu 3lOWh2Bo 192.168.4.108 33798
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv
<------------->
--- (13 headers 11 lines) ---
Sending to (Origination IP):8364 (NAT)
Using INVITE request as basis request - MGYzMDJiMTY3YmUzNGIwZDkyYjFjYTU2NjFmM2NiMTI.
Found peer '8001' for '8001' from (Origination IP):8364
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port (Origination IP):33798
Looking for 9919033016760 in default (domain (Vicidial Server IP))
list_route: hop: <sip:8001@(Origination IP):8364>

<--- Transmitting (NAT) to (Origination IP):8364 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP (Origination IP):8364;branch=z9hG4bK-d8754z-3b6e8b49f82c6334-1---d8754z-;received=(Origination IP);rport=8364
From: "8001"<sip:8001@(Vicidial Server IP)>;tag=f37ee657
To: "9919033016760"<sip:9919033016760@(Vicidial Server IP)>
Call-ID: MGYzMDJiMTY3YmUzNGIwZDkyYjFjYTU2NjFmM2NiMTI.
CSeq: 2 INVITE
Server: Asterisk PBX 11.22.0-vici
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:9919033016760@(Vicidial Server IP):5060>
Content-Length: 0


<------------>
Audio is at 11440
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to (SIP Carrier IP):5060:
INVITE sip:919033016760@(SIP Carrier IP) SIP/2.0
Via: SIP/2.0/UDP (Vicidial Server IP):5060;branch=z9hG4bK639c7771
Max-Forwards: 70
From: "Test Admin Phone" <sip:8001@(Vicidial Server IP)>;tag=as1a3fb598
To: <sip:919033016760@(SIP Carrier IP)>
Contact: <sip:8001@(Vicidial Server IP):5060>
Call-ID: 19d25481577dd6a06fbd982205d8e08d@(Vicidial Server IP):5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.22.0-vici
Date: Tue, 29 Aug 2017 16:14:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Remote-Party-ID: "Test Admin Phone" <sip:8001@(Vicidial Server IP)>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 267

v=0
o=root 1335175521 1335175521 IN IP4 (Vicidial Server IP)
s=Asterisk PBX 11.22.0-vici
c=IN IP4 (Vicidial Server IP)
t=0 0
m=audio 11440 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
Retransmitting #1 (no NAT) to (SIP Carrier IP):5060:
INVITE sip:919033016760@(SIP Carrier IP) SIP/2.0
Via: SIP/2.0/UDP (Vicidial Server IP):5060;branch=z9hG4bK639c7771
Max-Forwards: 70
From: "Test Admin Phone" <sip:8001@(Vicidial Server IP)>;tag=as1a3fb598
To: <sip:919033016760@(SIP Carrier IP)>
Contact: <sip:8001@(Vicidial Server IP):5060>
Call-ID: 19d25481577dd6a06fbd982205d8e08d@(Vicidial Server IP):5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.22.0-vici
Date: Tue, 29 Aug 2017 16:14:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Remote-Party-ID: "Test Admin Phone" <sip:8001@(Vicidial Server IP)>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 267

v=0
o=root 1335175521 1335175521 IN IP4 (Vicidial Server IP)
s=Asterisk PBX 11.22.0-vici
c=IN IP4 (Vicidial Server IP)
t=0 0
m=audio 11440 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:(SIP Carrier IP):5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP (Vicidial Server IP):5060;branch=z9hG4bK639c7771
From: "Test Admin Phone" <sip:8001@(Vicidial Server IP)>;tag=as1a3fb598
To: <sip:919033016760@(SIP Carrier IP)>
Contact: <sip:919033016760@(SIP Carrier IP):5060>
Call-Id: 19d25481577dd6a06fbd982205d8e08d@(Vicidial Server IP):5060
CSeq: 102 INVITE
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:(SIP Carrier IP):5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP (Vicidial Server IP):5060;branch=z9hG4bK639c7771
From: "Test Admin Phone" <sip:8001@(Vicidial Server IP)>;tag=as1a3fb598
To: <sip:919033016760@(SIP Carrier IP)>;tag=A382E24642836
Contact: <sip:919033016760@(SIP Carrier IP):5060>
Call-Id: 19d25481577dd6a06fbd982205d8e08d@(Vicidial Server IP):5060
CSeq: 102 INVITE
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:(SIP Carrier IP):5060 --->
SIP/2.0 408 Request Timeout
Via: SIP/2.0/UDP (Vicidial Server IP):5060;branch=z9hG4bK639c7771
From: "Test Admin Phone" <sip:8001@(Vicidial Server IP)>;tag=as1a3fb598
To: <sip:919033016760@(SIP Carrier IP)>
Call-Id: 19d25481577dd6a06fbd982205d8e08d@(Vicidial Server IP):5060
CSeq: 102 INVITE
Contact: <sip:8001@(SIP Carrier IP):5060>
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Transmitting (no NAT) to (SIP Carrier IP):5060:
ACK sip:919033016760@(SIP Carrier IP) SIP/2.0
Via: SIP/2.0/UDP (Vicidial Server IP):5060;branch=z9hG4bK639c7771
Max-Forwards: 70
From: "Test Admin Phone" <sip:8001@(Vicidial Server IP)>;tag=as1a3fb598
To: <sip:919033016760@(SIP Carrier IP)>
Contact: <sip:8001@(Vicidial Server IP):5060>
Call-ID: 19d25481577dd6a06fbd982205d8e08d@(Vicidial Server IP):5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.22.0-vici
Content-Length: 0


---
Scheduling destruction of SIP dialog '19d25481577dd6a06fbd982205d8e08d@(Vicidial Server IP):5060' in 6976 ms (Method: INVITE)
Scheduling destruction of SIP dialog 'MGYzMDJiMTY3YmUzNGIwZDkyYjFjYTU2NjFmM2NiMTI.' in 23872 ms (Method: INVITE)

<--- Reliably Transmitting (NAT) to (Origination IP):8364 --->
SIP/2.0 408 Request Timeout
Via: SIP/2.0/UDP (Origination IP):8364;branch=z9hG4bK-d8754z-3b6e8b49f82c6334-1---d8754z-;received=(Origination IP);rport=8364
From: "8001"<sip:8001@(Vicidial Server IP)>;tag=f37ee657
To: "9919033016760"<sip:9919033016760@(Vicidial Server IP)>;tag=as5170f41e
Call-ID: MGYzMDJiMTY3YmUzNGIwZDkyYjFjYTU2NjFmM2NiMTI.
CSeq: 2 INVITE
Server: Asterisk PBX 11.22.0-vici
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>

<--- SIP read from UDP:(Origination IP):8364 --->
ACK sip:9919033016760@(Vicidial Server IP) SIP/2.0
Via: SIP/2.0/UDP (Origination IP):8364;branch=z9hG4bK-d8754z-3b6e8b49f82c6334-1---d8754z-;rport
To: "9919033016760"<sip:9919033016760@(Vicidial Server IP)>;tag=as5170f41e
From: "8001"<sip:8001@(Vicidial Server IP)>;tag=f37ee657
Call-ID: MGYzMDJiMTY3YmUzNGIwZDkyYjFjYTU2NjFmM2NiMTI.
CSeq: 2 ACK
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

<--- SIP read from UDP:(Origination IP):8364 --->
INVITE sip:919033016760@(Vicidial Server IP) SIP/2.0
Via: SIP/2.0/UDP (Origination IP):8364;branch=z9hG4bK-d8754z-fb0059363c542123-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:8001@(Origination IP):8364>
To: "919033016760"<sip:919033016760@(Vicidial Server IP)>
From: "8001"<sip:8001@(Vicidial Server IP)>;tag=6f10f227
Call-ID: OWNjZjA3YzZiMGVhNjA4NTMxNjdiYzIzOGQ2ZjY4Yjc.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: eyeBeam release 1102q stamp 51814
Content-Length: 289

v=0
o=- 0 2 IN IP4 (Origination IP)
s=CounterPath eyeBeam 1.5
c=IN IP4 (Origination IP)
t=0 0
m=audio 17846 RTP/AVP 0 8 101
a=alt:1 2 : CduAGnNF BwQ58uaI 192.168.56.1 17846
a=alt:2 1 : HZCtnCip Upr9S2Td 192.168.4.108 17846
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv
<------------->
--- (12 headers 11 lines) ---
Sending to (Origination IP):8364 (NAT)
Sending to (Origination IP):8364 (NAT)
Using INVITE request as basis request - OWNjZjA3YzZiMGVhNjA4NTMxNjdiYzIzOGQ2ZjY4Yjc.
Found peer '8001' for '8001' from (Origination IP):8364

<--- Reliably Transmitting (NAT) to (Origination IP):8364 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP (Origination IP):8364;branch=z9hG4bK-d8754z-fb0059363c542123-1---d8754z-;received=(Origination IP);rport=8364
From: "8001"<sip:8001@(Vicidial Server IP)>;tag=6f10f227
To: "919033016760"<sip:919033016760@(Vicidial Server IP)>;tag=as580f715a
Call-ID: OWNjZjA3YzZiMGVhNjA4NTMxNjdiYzIzOGQ2ZjY4Yjc.
CSeq: 1 INVITE
Server: Asterisk PBX 11.22.0-vici
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="79dabccf"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'OWNjZjA3YzZiMGVhNjA4NTMxNjdiYzIzOGQ2ZjY4Yjc.' in 23872 ms (Method: INVITE)

<--- SIP read from UDP:(Origination IP):8364 --->
ACK sip:919033016760@(Vicidial Server IP) SIP/2.0
Via: SIP/2.0/UDP (Origination IP):8364;branch=z9hG4bK-d8754z-fb0059363c542123-1---d8754z-;rport
To: "919033016760"<sip:919033016760@(Vicidial Server IP)>;tag=as580f715a
From: "8001"<sip:8001@(Vicidial Server IP)>;tag=6f10f227
Call-ID: OWNjZjA3YzZiMGVhNjA4NTMxNjdiYzIzOGQ2ZjY4Yjc.
CSeq: 1 ACK
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

<--- SIP read from UDP:(Origination IP):8364 --->
INVITE sip:919033016760@(Vicidial Server IP) SIP/2.0
Via: SIP/2.0/UDP (Origination IP):8364;branch=z9hG4bK-d8754z-dc6c20536c345d37-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:8001@(Origination IP):8364>
To: "919033016760"<sip:919033016760@(Vicidial Server IP)>
From: "8001"<sip:8001@(Vicidial Server IP)>;tag=6f10f227
Call-ID: OWNjZjA3YzZiMGVhNjA4NTMxNjdiYzIzOGQ2ZjY4Yjc.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: eyeBeam release 1102q stamp 51814
Authorization: Digest username="8001",realm="asterisk",nonce="79dabccf",uri="sip:919033016760@(Vicidial Server IP)",response="a5fff30409b3f630751885e7de9f29b0",algorithm=MD5
Content-Length: 289

v=0
o=- 0 2 IN IP4 (Origination IP)
s=CounterPath eyeBeam 1.5
c=IN IP4 (Origination IP)
t=0 0
m=audio 17846 RTP/AVP 0 8 101
a=alt:1 2 : CduAGnNF BwQ58uaI 192.168.56.1 17846
a=alt:2 1 : HZCtnCip Upr9S2Td 192.168.4.108 17846
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv
<------------->
--- (13 headers 11 lines) ---
Sending to (Origination IP):8364 (NAT)
Using INVITE request as basis request - OWNjZjA3YzZiMGVhNjA4NTMxNjdiYzIzOGQ2ZjY4Yjc.
Found peer '8001' for '8001' from (Origination IP):8364
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port (Origination IP):17846
Looking for 919033016760 in default (domain (Vicidial Server IP))
list_route: hop: <sip:8001@(Origination IP):8364>

<--- Transmitting (NAT) to (Origination IP):8364 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP (Origination IP):8364;branch=z9hG4bK-d8754z-dc6c20536c345d37-1---d8754z-;received=(Origination IP);rport=8364
From: "8001"<sip:8001@(Vicidial Server IP)>;tag=6f10f227
To: "919033016760"<sip:919033016760@(Vicidial Server IP)>
Call-ID: OWNjZjA3YzZiMGVhNjA4NTMxNjdiYzIzOGQ2ZjY4Yjc.
CSeq: 2 INVITE
Server: Asterisk PBX 11.22.0-vici
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:919033016760@(Vicidial Server IP):5060>
Content-Length: 0


<------------>
Audio is at 19348
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to (SIP Carrier IP):5060:
INVITE sip:19033016760@(SIP Carrier IP) SIP/2.0
Via: SIP/2.0/UDP (Vicidial Server IP):5060;branch=z9hG4bK5d178aef
Max-Forwards: 70
From: "Test Admin Phone" <sip:8001@(Vicidial Server IP)>;tag=as21459d45
To: <sip:19033016760@(SIP Carrier IP)>
Contact: <sip:8001@(Vicidial Server IP):5060>
Call-ID: 6bc519a20efd130e15d805da1f41bd82@(Vicidial Server IP):5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.22.0-vici
Date: Tue, 29 Aug 2017 16:14:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Remote-Party-ID: "Test Admin Phone" <sip:8001@(Vicidial Server IP)>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 267

v=0
o=root 1567179260 1567179260 IN IP4 (Vicidial Server IP)
s=Asterisk PBX 11.22.0-vici
c=IN IP4 (Vicidial Server IP)
t=0 0
m=audio 19348 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
Retransmitting #1 (no NAT) to (SIP Carrier IP):5060:
INVITE sip:19033016760@(SIP Carrier IP) SIP/2.0
Via: SIP/2.0/UDP (Vicidial Server IP):5060;branch=z9hG4bK5d178aef
Max-Forwards: 70
From: "Test Admin Phone" <sip:8001@(Vicidial Server IP)>;tag=as21459d45
To: <sip:19033016760@(SIP Carrier IP)>
Contact: <sip:8001@(Vicidial Server IP):5060>
Call-ID: 6bc519a20efd130e15d805da1f41bd82@(Vicidial Server IP):5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.22.0-vici
Date: Tue, 29 Aug 2017 16:14:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Remote-Party-ID: "Test Admin Phone" <sip:8001@(Vicidial Server IP)>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 267

v=0
o=root 1567179260 1567179260 IN IP4 (Vicidial Server IP)
s=Asterisk PBX 11.22.0-vici
c=IN IP4 (Vicidial Server IP)
t=0 0
m=audio 19348 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:(SIP Carrier IP):5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP (Vicidial Server IP):5060;branch=z9hG4bK5d178aef
From: "Test Admin Phone" <sip:8001@(Vicidial Server IP)>;tag=as21459d45
To: <sip:19033016760@(SIP Carrier IP)>
Contact: <sip:19033016760@(SIP Carrier IP):5060>
Call-Id: 6bc519a20efd130e15d805da1f41bd82@(Vicidial Server IP):5060
CSeq: 102 INVITE
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:(SIP Carrier IP):5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP (Vicidial Server IP):5060;branch=z9hG4bK5d178aef
From: "Test Admin Phone" <sip:8001@(Vicidial Server IP)>;tag=as21459d45
To: <sip:19033016760@(SIP Carrier IP)>;tag=7B597913195L3
Contact: <sip:19033016760@(SIP Carrier IP):5060>
Call-Id: 6bc519a20efd130e15d805da1f41bd82@(Vicidial Server IP):5060
CSeq: 102 INVITE
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '19d25481577dd6a06fbd982205d8e08d@(Vicidial Server IP):5060' Method: INVITE

<--- SIP read from UDP:(SIP Carrier IP):5060 --->
SIP/2.0 408 Request Timeout
Via: SIP/2.0/UDP (Vicidial Server IP):5060;branch=z9hG4bK5d178aef
From: "Test Admin Phone" <sip:8001@(Vicidial Server IP)>;tag=as21459d45
To: <sip:19033016760@(SIP Carrier IP)>
Call-Id: 6bc519a20efd130e15d805da1f41bd82@(Vicidial Server IP):5060
CSeq: 102 INVITE
Contact: <sip:8001@(SIP Carrier IP):5060>
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Transmitting (no NAT) to (SIP Carrier IP):5060:
ACK sip:19033016760@(SIP Carrier IP) SIP/2.0
Via: SIP/2.0/UDP (Vicidial Server IP):5060;branch=z9hG4bK5d178aef
Max-Forwards: 70
From: "Test Admin Phone" <sip:8001@(Vicidial Server IP)>;tag=as21459d45
To: <sip:19033016760@(SIP Carrier IP)>
Contact: <sip:8001@(Vicidial Server IP):5060>
Call-ID: 6bc519a20efd130e15d805da1f41bd82@(Vicidial Server IP):5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.22.0-vici
Content-Length: 0


---
Scheduling destruction of SIP dialog '6bc519a20efd130e15d805da1f41bd82@(Vicidial Server IP):5060' in 6976 ms (Method: INVITE)
Scheduling destruction of SIP dialog 'OWNjZjA3YzZiMGVhNjA4NTMxNjdiYzIzOGQ2ZjY4Yjc.' in 23872 ms (Method: INVITE)

<--- Reliably Transmitting (NAT) to (Origination IP):8364 --->
SIP/2.0 408 Request Timeout
Via: SIP/2.0/UDP (Origination IP):8364;branch=z9hG4bK-d8754z-dc6c20536c345d37-1---d8754z-;received=(Origination IP);rport=8364
From: "8001"<sip:8001@(Vicidial Server IP)>;tag=6f10f227
To: "919033016760"<sip:919033016760@(Vicidial Server IP)>;tag=as4ea5831e
Call-ID: OWNjZjA3YzZiMGVhNjA4NTMxNjdiYzIzOGQ2ZjY4Yjc.
CSeq: 2 INVITE
Server: Asterisk PBX 11.22.0-vici
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>

<--- SIP read from UDP:(Origination IP):8364 --->
ACK sip:919033016760@(Vicidial Server IP) SIP/2.0
Via: SIP/2.0/UDP (Origination IP):8364;branch=z9hG4bK-d8754z-dc6c20536c345d37-1---d8754z-;rport
To: "919033016760"<sip:919033016760@(Vicidial Server IP)>;tag=as4ea5831e
From: "8001"<sip:8001@(Vicidial Server IP)>;tag=6f10f227
Call-ID: OWNjZjA3YzZiMGVhNjA4NTMxNjdiYzIzOGQ2ZjY4Yjc.
CSeq: 2 ACK
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

<--- SIP read from UDP:(Origination IP):8364 --->


<------------->

Re: Calls Failing With 408, Request Timeout.

PostPosted: Fri Sep 01, 2017 9:15 am
by kumar2arbind
This 408 is occur generally when the server could not produce a response within a suitable amount of time, are you server behind firewall or something else..
try by putting nat=yes in sip.conf

IS your soft phone register successfully.
and please share your cli log too.